1. Field of the Invention
The present invention relates generally to a device and method for supporting voice service in a mobile communication system, and in particular, to a device and method for communicating packet voice data, which can support more voice users.
2. Description of the Related Art
In a conventional mobile telephone system, as in a typical wire telephone service a fixed bandwidth is assigned from a voice call set-up to a voice call release in a line-type voice protocol. This corresponds to voice service over a mobile communication network such as IS-95, GSM (Global System for Mobile communication), and the like. When a call for voice service is established between a mobile station and a base station, fixed radio resources are assigned until the call is released. Therefore, the line-type voice protocol assigns fixed resources from call set-up to call release as shown in FIG. 1. A fixed assigned channel is assigned to a user even though the user does not continuously generate voice traffic, thereby preventing another user from using the channel.
Generally, voice traffic consists of an utterance period where sounds are produced and a mute period where no sounds are produced. While the ratio of utterance period to mute period varies with nation or individual user, research and analysis of user characteristics suggests that the ratio is 300 ms:700 ms, or 1 sec:1.35 sec.
A line-type voice service can be considered the best way to support voice quality because a fixed bandwidth is assigned all the time. From a user's perspective, however; the service is billed even for the mute period and thus the user pays for unused periods. From a service provider's perspective, bandwidth efficiency is decreased by assigning a fixed radio resource, the fixed radio resource being a very small bandwidth, as compared to a wire service.
Therefore, line-type service structure needs to change to a packet type so that other activated users can use the bandwidth. Theoretically, in the case of utterance period:mute period=300 ms:700 ms, three times more subscribers can be supported, and thus performance can be increased drastically.
However, slow progress has been made in this area because the existing radio resource managing scheme requires a large time delay and makes real-time control difficult. This is particularly true in IS-95, a control method which uses a 20 ms-control message. An IS-95 system performs control processes on a common channel and then multiplexes signal traffic and voice traffic on one channel by inband signaling in providing a service. Therefore, in order to support a mechanism of releasing a traffic channel during a mute period and assigning it during an utterance period, the traffic channel should be acquired through the common channel when the utterance period is entered. In this case, a time delay occurs due to contention-based channel acquisition. In addition, if the traffic channel is released during the mute period, control information cannot be transmitted because of inband signaling. Furthermore, since various functions including power control are related to the operation of a traffic channel, the traffic channel cannot be dynamically assigned and released in the conventional technology. Use of a 20 ms-control message incurs a delay of a few hundred milliseconds because of the processing time in requesting assignment of a traffic channel and releasing it in an active state. Hence, voice service quality cannot be ensured.
There are other conventional packet voice protocols in a wire network. They utilize the fact that a voice service has a mute period and an utterance period, to thereby efficiently use limited bandwidths. While connection-type line technology bills on the basis of time, the packet voice protocols bill a user on the basis of information about actual use of the network by packet-unit billing.
The packet voice services which have been studied so far have been designed and developed for use in wire networks. Discussion has been made about supporting this service over a radio network but no specifics have been suggested yet. This is because the structure of current mobile phone service cannot support packet-based technology.
The current packet voice service is based on ITU-T (International Telecommunication Union) G.764 “Packetized Voice Protocol”. This is a LAN technology designed for widely used common channel access schemes but utilizable over a wire communication network. An Internet phone, which is a very popular packet voice service on the Internet, is designed based on the above technology. The difference between the Internet phone and the ITU-T G.764 is that the former is a layer-2 protocol while the latter uses RTP/RTCP (Realtime Transmission Protocol/Realtime Transmission Control Protocol) of the IETF (Internet Engineering Task Force) and is designed as a layer-4 protocol. The Internet phone is intended for use as part of a TCP/IP (Transmission Control Protocol/Internet Protocol) network since the RTP/RTCP is designed based on the IETF, and to efficiently use an existing IP network. However, the Internet phone and the ITU-T G.764 are almost the same in operation.
A problem with extension of the ITU-T G.764 to a radio network is that conventional wire network-based packet voice protocols such as the ITU-T G.764 are characterized by contention-based use of common channels. That is, over the Internet or a LAN, common channels are used and traffic transmission is implemented on the basis of contention between users (see FIG. 2). Therefore, no separate channel reservation technique for acquisition of a common channel and no channel release technique are necessary. In addition, the ability of the transmission end to detect the presence or absence of a contention is a feature of communication in non-connection service over a wire network.
When a common channel is used without a reservation in a radio communication network, the contention-caused time delay may have a great influence. It is impossible to detect a contention in the radio environment and a contention-based scheme shows a poor performance. Therefore, it is very difficult to design a radio packet voice protocol using a contention-based scheme. In particular, considering CDMA technology, power control and synchronization make it impossible to support a contention-based packet voice protocol using common channels.
Furthermore, the radio mobile communication network requires transmission/reception signals and control information to transmit voice traffic, maintain a call, and exchange control information. Thus, the common channel-based contention scheme is difficult to support. Especially, handoff of a mobile station makes it more difficult to support because time delay and inconvenience are involved in transmission/reception of handoff-related information and process messages.
Therefore, the packet voice protocols like ITU-T G.764 or the IETF RTP/RTCP, which were designed based on the conventional wire network, are difficult to use and inefficient in a radio channel environment using CDMA technology.